AstRecipes » Using a HT-488 with Asterisk

The Grandstream HandyTone HT-488 is a low-cost SIP analog adapter that offers one FXS port and one FXO port to connect one analog telephone and one POTS line to the unit.

The scenario
We want to install the unit with an Asterisk server whick IP is 10.10.3.5 and want to register the telephone attached to the unit as SIP/32, while the FXO line will be seen by Asterisk as SIP/33.

Installing the HT-488
As it ships, access for the web interface is disabled for users accessing from the WAN. To enable it, connect the LAN port of the unit to a PC with a cross-cable patch, use the unit's own DHCP server to assign address to the PC and logon by navigating to http://192.168.2.1 (password admin) - go to Basic settings and set WAN side http access. Save and reboot.

Connect the unit to your LAN through the WAN port and connect to it through its DHCP-assigned address (to see what that is, connect a phone to the FXS port and digit ***02 to have the unit read out the assigned IP address). If the unit did not get an IP address, select the option ***01 and then 9 to switch from fixed-ip to DHCP mode.

Firmware upgrade
It is VERY IMPORTANT that you set the unit with a modern firmware, or you'll have a number of problems. Luckily this is very easy: check the current TFTP server address on Grandsteram site (see below) and enter it under the configuration page, and set "Yes, check for upgrade every 1 minute". Reboot.
The unit light should go red while the unit downloads and installs the new firmware, it will take like 5 minutes to update.

The Forward to Voip option must be set to "s" in order to have incoming FXO calls sent over to the Asterisk server.

FXS PORT

SIP Server: 10.10.3.5

Outbound Proxy: 10.10.3.5

SIP User ID: 32

Authenticate ID: 32

Authenticate Password: XXX

Name: (leave blank)

SIP Registration: yes

Unregister On Reboot: yes

Send DTMF: via RTP (RFC2833)

Send Flash Event: Yes

Enable Call Features: No

NAT Traversal (STUN): No

No Key Entry Timeout: 4

Preferred Vocoder: PCMU / PCMA

Silence Suppression: No

Fax Mode: Pass-through

FXO PORT

SIP Server: 10.10.3.5

Outbound Proxy: 10.10.3.5

SIP User ID: 33

Authenticate ID: 33

Authenticate Password: XXX

Name: (leave blank)

SIP Registration: yes

Unregister On Reboot: yes

Send DTMF: via RTP (RFC2833)

Send Flash Event: No

NAT Traversal (STUN): No

Preferred Vocoder: PCMU / PCMA

Silence Suppression: No

Fax-mode: Pass-through

PSTN AC Termination: 270 Ohm

Enable PSTN Disconnect Tone Detection: yes

PSTN Silence Timeout: 20

Enable Current Disconnect: Yes

Save and reboot

Asterisk configuration

In SIP.conf enter the following piece of code:

; FXS port

[32]

type=friend

secret=XXX

callerid="My FXS" <32>

host=dynamic

nat=no

canreinvite=no

disallow=all

allow=ulaw

context=sip

qualify=yes

dtmfmode=rfc2833

; FXO port

[33]

type=friend

secret=XXX

callerid="My FXO" <33>

host=dynamic

nat=no

canreinvite=no

disallow=all

allow=ulaw

context=in-sip

qualify=yes

dtmfmode=rfc2833

Once the phones are registered, you have a standard SIP phone as SIP/32 that will send calls to the context sip, while the SIP/33 POTS line can be dialled as Dial(SIP/33/number,30) and will send incoming calls to s@in-sip.